Calls failing in queue

tobyLIT · August 17, 2022, 6:01am

Hi.

I’ve got some calls that appear to failing in the queue and I can’t figured it out.

What exactly does the different Dispositions mean? i.e. Answered, Busy, Failed and No Answer

david55 · August 17, 2022, 9:41am

https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification#Asterisk12CDRSpecification-Dispositions

tobyLIT · August 18, 2022, 4:25am

So I looked at it and it’s the same DID and Destination 9800 (Queue)

tobyLIT · August 18, 2022, 4:29am

@david55 Where could I find the hangup cause like which log file would they show up in?

david55 · August 18, 2022, 10:10am

My knowledge is of Aserisk, rather than FreePBX, but it looks like it will appear in the verbose logging of the dialplan, although it looks like you only get it reported explicitly for dialing trunks:

github.com

FreePBX/core/blob/cff74910046d40dc6eeb9b4ef01664144f7d0711/functions.inc.php#L2594


      
          			$ext->add($context, $exten, '', new ext_execif('$[$["${DB(AMPUSER/${AMPUSER}/cidname)}" != ""] & $["${CALLERID(name)}"="hidden"]]', 'Set', 'CONNECTEDLINE(name,i)=CID:(Hidden)${CALLERID(number)}'));
          		}
          
          
		$ext->add($context, $exten, '', new ext_gotoif('$["${custom}" = "AMP"]', 'customtrunk'));
          		$ext->add($context, $exten, '', new ext_execif('$["${DIRECTION}" = "INBOUND"]', 'Set', 'DIAL_TRUNK_OPTIONS=${STRREPLACE(DIAL_TRUNK_OPTIONS,T)}'));
          		$ext->add($context, $exten, '', new ext_set("HASH(__SIPHEADERS,Alert-Info)", "unset"));
          
          
		$obroute_email = ($route['notification_on'] == 'call') ? 'U(sub-send-obroute-email^${DIAL_NUMBER}^${MACRO_EXTEN}^${DIAL_TRUNK}^${NOW}^${CALLERID(name)}^${CALLERID(number)})' : '';
          
          
		$ext->add($context, $exten, '', new ext_dial('${OUT_${DIAL_TRUNK}}/${OUTNUM}${OUT_${DIAL_TRUNK}_SUFFIX}', '${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}b(func-apply-sipheaders^s^1,(${DIAL_TRUNK}))'.$obroute_email));  // Regular Trunk Dial
          		$ext->add($context, $exten, '', new ext_noop('Dial failed for some reason with DIALSTATUS = ${DIALSTATUS} and HANGUPCAUSE = ${HANGUPCAUSE}'));
          		$ext->add($context, $exten, '', new ext_gotoif('$["${ARG4}" = "on"]','continue,1', 's-${DIALSTATUS},1'));
          
          
		$ext->add($context, $exten, 'customtrunk', new ext_set('pre_num', '${CUT(OUT_${DIAL_TRUNK},$,1)}'));
          		$ext->add($context, $exten, '', new ext_set('the_num', '${CUT(OUT_${DIAL_TRUNK},$,2)}'));  // this is where we expect to find string OUTNUM
          		$ext->add($context, $exten, '', new ext_set('post_num', '${CUT(OUT_${DIAL_TRUNK},$,3)}'));
          		$ext->add($context, $exten, '', new ext_gotoif('$["${the_num}" = "OUTNUM"]', 'outnum', 'skipoutnum'));  // if we didn't find "OUTNUM", then skip to Dial
          		$ext->add($context, $exten, 'outnum', new ext_set('the_num', '${OUTNUM}'));  // replace "OUTNUM" with the actual number to dial
          		$ext->add($context, $exten, 'skipoutnum', new ext_dial('${pre_num:4}${the_num}${post_num}', '${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}'));
          		$ext->add($context, $exten, '', new ext_noop('Dial failed for some reason with DIALSTATUS = ${DIALSTATUS} and HANGUPCAUSE = ${HANGUPCAUSE}'));
          		$ext->add($context, $exten, '', new ext_gotoif('$["${ARG4}" = "on"]','continue,1', 's-${DIALSTATUS},1'));

It’s generally derived from the signalling protocol. It should be literally what appears in the signalling for ISDN. Asterisk uses a slightly modified version of the table in:

https://www.rfc-editor.org/rfc/rfc3398.html#section-8.2.6.1

to translate SIP causes, and you can see SIP causes, for chan_pjsip, in the verbose or full logging, after using the CLI command “pjsip set logger on”.

david55 · August 18, 2022, 12:23pm

I should have added that the key to the cause numbers is here:

github.com

asterisk/asterisk/blob/master/include/asterisk/causes.h#L39


      
          ${HANGUPCAUSE} channel variable after a call (after execution
          of "dial").
          
          
In SIP, we have a conversion table to convert between SIP
          return codes and Q.931 both ways. This is to improve SIP/ISDN
          compatibility.
          
          
These are the current codes, based on the Q.850/Q.931
          specification:
          
          
	- AST_CAUSE_UNALLOCATED                      1
          	- AST_CAUSE_NO_ROUTE_TRANSIT_NET             2
          	- AST_CAUSE_NO_ROUTE_DESTINATION             3
          	- AST_CAUSE_MISDIALLED_TRUNK_PREFIX          5
          	- AST_CAUSE_CHANNEL_UNACCEPTABLE             6
          	- AST_CAUSE_CALL_AWARDED_DELIVERED           7
          	- AST_CAUSE_PRE_EMPTED                       8
          	- AST_CAUSE_NUMBER_PORTED_NOT_HERE          14
          	- AST_CAUSE_NORMAL_CLEARING                 16
          	- AST_CAUSE_USER_BUSY                       17
          	- AST_CAUSE_NO_USER_RESPONSE                18

tobyLIT · August 22, 2022, 2:20am

I’m confused exactly where I should be looking, where are the dialplan log files? or what do you mean dialing trunks

Sorry for not knowing these things as I’m quite new when it comes to asterisk/freebpx

david55 · August 22, 2022, 9:42am

That should have been pjsip set logger on.

The log files are in /var/log/asterisk/, although I think there is also GUI access to them. See Providing Great Debug - Support Services - Documentation for the information anyone debugging FreePBX ought to know.

QM2021 · August 24, 2022, 4:54am

I’d start by having a look at what gets written on the queue_log for those specific calls.

tobyLIT · August 28, 2022, 11:25pm

So I’ve had a look at the queue_log and found that there is a quite a few ‘ABANDON’ so I looked at Queue Logs - Asterisk Project - Asterisk Project Wiki and found that the caller abandoned their position in the queue. Does that mean that they hanged up? or did something happen to their connection?

david55 · August 28, 2022, 11:31pm

It could mean either, but one would normally assume the caller hung up.

QM2021 · August 29, 2022, 7:12am

Definitely. 99.9% of the cases it was hung up by the caller. If you are unsure, e.g. because you get a spike, you can find the unique-id of such calls and go look on the main Asterisk log for something fishy.

tobyLIT · August 31, 2022, 3:33am

I was hopping that it was just the caller hanging up but I don’t think that is the case as the duration of the call is 0:00.

QM2021 · September 5, 2022, 9:36am

You need to have a look at the main log, you can grep it by call-id so you’ll see what is happening at the call level.

tobyLIT · September 11, 2022, 11:21pm

I finally got time to take a look at it and I’m not exactly sure what I should grep in the main log but looking at the CDR reports I can see that a number shows up multiple times if you look at the picture you can see that this all happens in 1 second - the caller ID stays the same throughout and the DID also stays the same.

tobyLIT · September 11, 2022, 11:26pm

To if I look up that number I find that it’s a vineyard number, could it be a spam caller?

QM2021 · September 12, 2022, 5:11am

Sending you tens if not hundreds of calls in such a short period of time? to me it looks more like a routing issue - calls joins the queue, is kicked out, and is joined again.

PitzKey · September 12, 2022, 5:25am

Post a call trace via pastebin: https://wiki.freepbx.org/display/SUP/Providing+Great+Debug#ProvidingGreatDebug-AsteriskLogs-PartII

Also, grep the unique ID from the queue log to see why it exited the queue

system · October 13, 2022, 5:26am

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